mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
fix #133, support push rtsp to srs. 2.0.120.
This commit is contained in:
winlin
parent
a954040d29
commit
9d233db27e
11 changed files with 670 additions and 80 deletions
|
|
@ -35,6 +35,12 @@ using namespace std;
|
|||
#include <srs_core_autofree.hpp>
|
||||
#include <srs_kernel_stream.hpp>
|
||||
#include <srs_kernel_buffer.hpp>
|
||||
#include <srs_rtmp_sdk.hpp>
|
||||
#include <srs_rtmp_amf0.hpp>
|
||||
#include <srs_rtmp_utility.hpp>
|
||||
#include <srs_kernel_utility.hpp>
|
||||
#include <srs_raw_avc.hpp>
|
||||
#include <srs_kernel_codec.hpp>
|
||||
|
||||
#ifdef SRS_AUTO_STREAM_CASTER
|
||||
|
||||
|
|
@ -100,15 +106,15 @@ int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
|
|||
}
|
||||
}
|
||||
|
||||
srs_trace("rtsp: rtp %dB, vt=%d/%u, sts=%u/%#x/%#x, paylod=%dB, chunked=%d",
|
||||
nb_buf, cache->version, cache->payload_type, cache->sequence_number, cache->timestamp, cache->ssrc,
|
||||
srs_trace("rtsp: rtp #%d %dB, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB, chunked=%d",
|
||||
stream_id, nb_buf, cache->version, cache->payload_type, cache->sequence_number, cache->timestamp, cache->ssrc,
|
||||
cache->payload->length(), cache->chunked
|
||||
);
|
||||
|
||||
// always free it.
|
||||
SrsAutoFree(SrsRtpPacket, cache);
|
||||
|
||||
if ((ret = rtsp->on_rtp_packet(cache)) != ERROR_SUCCESS) {
|
||||
if ((ret = rtsp->on_rtp_packet(cache, stream_id)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: process rtp packet failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
|
|
@ -116,9 +122,59 @@ int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
|
|||
return ret;
|
||||
}
|
||||
|
||||
SrsRtspAudioCache::SrsRtspAudioCache()
|
||||
{
|
||||
dts = NULL;
|
||||
audio_samples = NULL;
|
||||
payload = NULL;
|
||||
}
|
||||
|
||||
SrsRtspAudioCache::~SrsRtspAudioCache()
|
||||
{
|
||||
srs_freep(audio_samples);
|
||||
srs_freep(payload);
|
||||
}
|
||||
|
||||
SrsRtspJitter::SrsRtspJitter()
|
||||
{
|
||||
delta = 0;
|
||||
previous_timestamp = 0;
|
||||
pts = 0;
|
||||
}
|
||||
|
||||
SrsRtspJitter::~SrsRtspJitter()
|
||||
{
|
||||
}
|
||||
|
||||
int64_t SrsRtspJitter::timestamp()
|
||||
{
|
||||
return pts;
|
||||
}
|
||||
|
||||
int SrsRtspJitter::correct(int64_t& ts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
if (previous_timestamp == 0) {
|
||||
previous_timestamp = ts;
|
||||
}
|
||||
|
||||
delta = srs_max(0, ts - previous_timestamp);
|
||||
if (delta > 90000) {
|
||||
delta = 0;
|
||||
}
|
||||
|
||||
previous_timestamp = ts;
|
||||
|
||||
ts = pts + delta;
|
||||
pts = ts;
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
|
||||
{
|
||||
output = o;
|
||||
output_template = o;
|
||||
|
||||
session = "";
|
||||
video_rtp = NULL;
|
||||
|
|
@ -129,6 +185,18 @@ SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
|
|||
skt = new SrsStSocket(fd);
|
||||
rtsp = new SrsRtspStack(skt);
|
||||
trd = new SrsThread("rtsp", this, 0, false);
|
||||
|
||||
req = NULL;
|
||||
io = NULL;
|
||||
client = NULL;
|
||||
stream_id = 0;
|
||||
vjitter = new SrsRtspJitter();
|
||||
ajitter = new SrsRtspJitter();
|
||||
|
||||
avc = new SrsRawH264Stream();
|
||||
aac = new SrsRawAacStream();
|
||||
acodec = new SrsRawAacStreamCodec();
|
||||
acache = new SrsRtspAudioCache();
|
||||
}
|
||||
|
||||
SrsRtspConn::~SrsRtspConn()
|
||||
|
|
@ -142,6 +210,13 @@ SrsRtspConn::~SrsRtspConn()
|
|||
srs_freep(trd);
|
||||
srs_freep(skt);
|
||||
srs_freep(rtsp);
|
||||
|
||||
close();
|
||||
|
||||
srs_freep(vjitter);
|
||||
srs_freep(ajitter);
|
||||
srs_freep(acodec);
|
||||
srs_freep(acache);
|
||||
}
|
||||
|
||||
int SrsRtspConn::serve()
|
||||
|
|
@ -179,6 +254,18 @@ int SrsRtspConn::do_cycle()
|
|||
return ret;
|
||||
}
|
||||
} else if (req->is_announce()) {
|
||||
if (rtsp_tcUrl.empty()) {
|
||||
rtsp_tcUrl = req->uri;
|
||||
}
|
||||
size_t pos = string::npos;
|
||||
if ((pos = rtsp_tcUrl.rfind(".sdp")) != string::npos) {
|
||||
rtsp_tcUrl = rtsp_tcUrl.substr(0, pos);
|
||||
}
|
||||
if ((pos = rtsp_tcUrl.rfind("/")) != string::npos) {
|
||||
rtsp_stream = rtsp_tcUrl.substr(pos + 1);
|
||||
rtsp_tcUrl = rtsp_tcUrl.substr(0, pos);
|
||||
}
|
||||
|
||||
srs_assert(req->sdp);
|
||||
video_id = ::atoi(req->sdp->video_stream_id.c_str());
|
||||
audio_id = ::atoi(req->sdp->audio_stream_id.c_str());
|
||||
|
|
@ -186,13 +273,13 @@ int SrsRtspConn::do_cycle()
|
|||
audio_codec = req->sdp->audio_codec;
|
||||
audio_sample_rate = ::atoi(req->sdp->audio_sample_rate.c_str());
|
||||
audio_channel = ::atoi(req->sdp->audio_channel.c_str());
|
||||
sps = req->sdp->video_sps;
|
||||
pps = req->sdp->video_pps;
|
||||
asc = req->sdp->audio_sh;
|
||||
srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels)",
|
||||
h264_sps = req->sdp->video_sps;
|
||||
h264_pps = req->sdp->video_pps;
|
||||
aac_specific_config = req->sdp->audio_sh;
|
||||
srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels), %s/%s",
|
||||
video_id, video_codec.c_str(), req->sdp->video_protocol.c_str(), req->sdp->video_transport_format.c_str(),
|
||||
audio_id, audio_codec.c_str(), req->sdp->audio_protocol.c_str(), req->sdp->audio_transport_format.c_str(),
|
||||
audio_sample_rate, audio_channel
|
||||
audio_sample_rate, audio_channel, rtsp_tcUrl.c_str(), rtsp_stream.c_str()
|
||||
);
|
||||
|
||||
SrsRtspResponse* res = new SrsRtspResponse(req->seq);
|
||||
|
|
@ -262,9 +349,38 @@ int SrsRtspConn::do_cycle()
|
|||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::on_rtp_packet(SrsRtpPacket* pkt)
|
||||
int SrsRtspConn::on_rtp_packet(SrsRtpPacket* pkt, int stream_id)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// ensure rtmp connected.
|
||||
if ((ret = connect()) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
if (stream_id == video_id) {
|
||||
// rtsp tbn is ts tbn.
|
||||
int64_t pts = pkt->timestamp;
|
||||
if ((ret = vjitter->correct(pts)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: correct by jitter failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
// TODO: FIXME: set dts to pts, please finger out the right dts.
|
||||
int64_t dts = pts;
|
||||
|
||||
return on_rtp_video(pkt, dts, pts);
|
||||
} else {
|
||||
// rtsp tbn is ts tbn.
|
||||
int64_t pts = pkt->timestamp;
|
||||
if ((ret = ajitter->correct(pts)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: correct by jitter failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return on_rtp_audio(pkt, pts);
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -307,6 +423,336 @@ void SrsRtspConn::on_thread_stop()
|
|||
caster->remove(this);
|
||||
}
|
||||
|
||||
int SrsRtspConn::on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
if ((ret = kickoff_audio_cache(pkt, dts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
if ((ret = write_h264_ipb_frame(pkt->payload->bytes(), pkt->payload->length(), dts / 90, pts / 90)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::on_rtp_audio(SrsRtpPacket* pkt, int64_t dts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
if ((ret = kickoff_audio_cache(pkt, dts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// cache current audio to kickoff.
|
||||
acache->dts = dts;
|
||||
acache->audio_samples = pkt->audio_samples;
|
||||
acache->payload = pkt->payload;
|
||||
|
||||
pkt->audio_samples = NULL;
|
||||
pkt->payload = NULL;
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// nothing to kick off.
|
||||
if (!acache->payload) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
if (dts - acache->dts > 0 && acache->audio_samples->nb_sample_units > 0) {
|
||||
int64_t delta = (dts - acache->dts) / acache->audio_samples->nb_sample_units;
|
||||
for (int i = 0; i < acache->audio_samples->nb_sample_units; i++) {
|
||||
char* frame = acache->audio_samples->sample_units[i].bytes;
|
||||
int nb_frame = acache->audio_samples->sample_units[i].size;
|
||||
int64_t timestamp = (acache->dts + delta * i) / 90;
|
||||
acodec->aac_packet_type = 1;
|
||||
if ((ret = write_audio_raw_frame(frame, nb_frame, acodec, timestamp)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
acache->dts = 0;
|
||||
srs_freep(acache->audio_samples);
|
||||
srs_freep(acache->payload);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::write_sequence_header()
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// use the current dts.
|
||||
int64_t dts = vjitter->timestamp() / 90;
|
||||
|
||||
// send video sps/pps
|
||||
if ((ret = write_h264_sps_pps(dts, dts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// generate audio sh by audio specific config.
|
||||
if (true) {
|
||||
std::string sh = aac_specific_config;
|
||||
|
||||
SrsAvcAacCodec dec;
|
||||
if ((ret = dec.audio_aac_sequence_header_demux((char*)sh.c_str(), (int)sh.length())) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
acodec->sound_format = SrsCodecAudioAAC;
|
||||
acodec->sound_type = (dec.aac_channels == 2)? SrsCodecAudioSoundTypeStereo : SrsCodecAudioSoundTypeMono;
|
||||
acodec->sound_size = SrsCodecAudioSampleSize16bit;
|
||||
acodec->aac_packet_type = 0;
|
||||
|
||||
static int aac_sample_rates[] = {
|
||||
96000, 88200, 64000, 48000,
|
||||
44100, 32000, 24000, 22050,
|
||||
16000, 12000, 11025, 8000,
|
||||
7350, 0, 0, 0
|
||||
};
|
||||
switch (aac_sample_rates[dec.aac_sample_rate]) {
|
||||
case 11025:
|
||||
acodec->sound_rate = SrsCodecAudioSampleRate11025;
|
||||
break;
|
||||
case 22050:
|
||||
acodec->sound_rate = SrsCodecAudioSampleRate22050;
|
||||
break;
|
||||
case 44100:
|
||||
acodec->sound_rate = SrsCodecAudioSampleRate44100;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
};
|
||||
|
||||
if ((ret = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), acodec, dts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// h264 raw to h264 packet.
|
||||
std::string sh;
|
||||
if ((ret = avc->mux_sequence_header(h264_sps, h264_pps, dts, pts, sh)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// h264 packet to flv packet.
|
||||
int8_t frame_type = SrsCodecVideoAVCFrameKeyFrame;
|
||||
int8_t avc_packet_type = SrsCodecVideoAVCTypeSequenceHeader;
|
||||
char* flv = NULL;
|
||||
int nb_flv = 0;
|
||||
if ((ret = avc->mux_avc2flv(sh, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// the timestamp in rtmp message header is dts.
|
||||
u_int32_t timestamp = dts;
|
||||
if ((ret = rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsRtspConn::write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
std::string ibp;
|
||||
int8_t frame_type;
|
||||
if ((ret = avc->mux_ipb_frame(frame, frame_size, dts, pts, ibp, frame_type)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
int8_t avc_packet_type = SrsCodecVideoAVCTypeNALU;
|
||||
char* flv = NULL;
|
||||
int nb_flv = 0;
|
||||
if ((ret = avc->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// the timestamp in rtmp message header is dts.
|
||||
u_int32_t timestamp = dts;
|
||||
return rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv);
|
||||
}
|
||||
|
||||
int SrsRtspConn::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
char* data = NULL;
|
||||
int size = 0;
|
||||
if ((ret = aac->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
return rtmp_write_packet(SrsCodecFlvTagAudio, dts, data, size);
|
||||
}
|
||||
|
||||
int SrsRtspConn::rtmp_write_packet(char type, u_int32_t timestamp, char* data, int size)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
SrsSharedPtrMessage* msg = NULL;
|
||||
|
||||
if ((ret = srs_rtmp_create_msg(type, timestamp, data, size, stream_id, &msg)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: create shared ptr msg failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
srs_assert(msg);
|
||||
|
||||
// send out encoded msg.
|
||||
if ((ret = client->send_and_free_message(msg, stream_id)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
// TODO: FIXME: merge all client code.
|
||||
int SrsRtspConn::connect()
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// when ok, ignore.
|
||||
if (io || client) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// parse uri
|
||||
if (!req) {
|
||||
req = new SrsRequest();
|
||||
|
||||
std::string schema, host, vhost, app, port, param;
|
||||
srs_discovery_tc_url(rtsp_tcUrl, schema, host, vhost, app, port, param);
|
||||
|
||||
// generate output by template.
|
||||
std::string output = output_template;
|
||||
output = srs_string_replace(output, "[app]", app);
|
||||
output = srs_string_replace(output, "[stream]", rtsp_stream);
|
||||
|
||||
size_t pos = string::npos;
|
||||
string uri = req->tcUrl = output;
|
||||
|
||||
// tcUrl, stream
|
||||
if ((pos = uri.rfind("/")) != string::npos) {
|
||||
req->stream = uri.substr(pos + 1);
|
||||
req->tcUrl = uri = uri.substr(0, pos);
|
||||
}
|
||||
|
||||
srs_discovery_tc_url(req->tcUrl,
|
||||
req->schema, req->host, req->vhost, req->app, req->port,
|
||||
req->param);
|
||||
}
|
||||
|
||||
// connect host.
|
||||
if ((ret = srs_socket_connect(req->host, ::atoi(req->port.c_str()), ST_UTIME_NO_TIMEOUT, &stfd)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect server %s:%s failed. ret=%d", req->host.c_str(), req->port.c_str(), ret);
|
||||
return ret;
|
||||
}
|
||||
io = new SrsStSocket(stfd);
|
||||
client = new SrsRtmpClient(io);
|
||||
|
||||
client->set_recv_timeout(SRS_CONSTS_RTMP_RECV_TIMEOUT_US);
|
||||
client->set_send_timeout(SRS_CONSTS_RTMP_SEND_TIMEOUT_US);
|
||||
|
||||
// connect to vhost/app
|
||||
if ((ret = client->handshake()) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: handshake with server failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
if ((ret = connect_app(req->host, req->port)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
if ((ret = client->create_stream(stream_id)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed, stream_id=%d. ret=%d", stream_id, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
// publish.
|
||||
if ((ret = client->publish(req->stream, stream_id)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: publish failed, stream=%s, stream_id=%d. ret=%d",
|
||||
req->stream.c_str(), stream_id, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return write_sequence_header();
|
||||
}
|
||||
|
||||
// TODO: FIXME: refine the connect_app.
|
||||
int SrsRtspConn::connect_app(string ep_server, string ep_port)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// args of request takes the srs info.
|
||||
if (req->args == NULL) {
|
||||
req->args = SrsAmf0Any::object();
|
||||
}
|
||||
|
||||
// notify server the edge identity,
|
||||
// @see https://github.com/winlinvip/simple-rtmp-server/issues/147
|
||||
SrsAmf0Object* data = req->args;
|
||||
data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
|
||||
data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
|
||||
data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
|
||||
data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
|
||||
data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));
|
||||
data->set("srs_version", SrsAmf0Any::str(RTMP_SIG_SRS_VERSION));
|
||||
data->set("srs_site", SrsAmf0Any::str(RTMP_SIG_SRS_WEB));
|
||||
data->set("srs_email", SrsAmf0Any::str(RTMP_SIG_SRS_EMAIL));
|
||||
data->set("srs_copyright", SrsAmf0Any::str(RTMP_SIG_SRS_COPYRIGHT));
|
||||
data->set("srs_primary", SrsAmf0Any::str(RTMP_SIG_SRS_PRIMARY));
|
||||
data->set("srs_authors", SrsAmf0Any::str(RTMP_SIG_SRS_AUTHROS));
|
||||
// for edge to directly get the id of client.
|
||||
data->set("srs_pid", SrsAmf0Any::number(getpid()));
|
||||
data->set("srs_id", SrsAmf0Any::number(_srs_context->get_id()));
|
||||
|
||||
// local ip of edge
|
||||
std::vector<std::string> ips = srs_get_local_ipv4_ips();
|
||||
assert(_srs_config->get_stats_network() < (int)ips.size());
|
||||
std::string local_ip = ips[_srs_config->get_stats_network()];
|
||||
data->set("srs_server_ip", SrsAmf0Any::str(local_ip.c_str()));
|
||||
|
||||
// generate the tcUrl
|
||||
std::string param = "";
|
||||
std::string tc_url = srs_generate_tc_url(ep_server, req->vhost, req->app, ep_port, param);
|
||||
|
||||
// upnode server identity will show in the connect_app of client.
|
||||
// @see https://github.com/winlinvip/simple-rtmp-server/issues/160
|
||||
// the debug_srs_upnode is config in vhost and default to true.
|
||||
bool debug_srs_upnode = _srs_config->get_debug_srs_upnode(req->vhost);
|
||||
if ((ret = client->connect_app(req->app, tc_url, req, debug_srs_upnode)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed, tcUrl=%s, dsu=%d. ret=%d",
|
||||
tc_url.c_str(), debug_srs_upnode, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
void SrsRtspConn::close()
|
||||
{
|
||||
srs_freep(client);
|
||||
srs_freep(io);
|
||||
srs_freep(req);
|
||||
srs_close_stfd(stfd);
|
||||
}
|
||||
|
||||
SrsRtspCaster::SrsRtspCaster(SrsConfDirective* c)
|
||||
{
|
||||
// TODO: FIXME: support reload.
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue